asterisk users - Apr 2009 - Asterisk-beginner : cannot make phonecalls using Asterisk

Hi there,

this is the first time that I'm building an Asterisk-server.

I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.

Thought it would go easier...

I have 2 Grandstream IP-phones : BT-201 and GXP-1200.

These are my settings :

sip.conf :

[root at asterisk asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr = 0.0.0.0

[BT201]
type=friend
context=intern
host=192.168.4.210
secret=testpaswoord

[GXP1200]
type=friend
context=intern
host=192.168.4.211
secret=testpaswoord

extensions.conf :

[root at asterisk asterisk]# cat extensions.conf
[intern]
exten => 210,1,Dial(SIP/BT201)
exten => 211,1,Dial(SIP/GXP1200)

Asterisk CLI shows me :

asterisk*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
             
GXP1200                    192.168.4.211               5060
 Unmonitored           
BT201                      192.168.4.210               5060
 Unmonitored           
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]

asterisk*CLI> dialplan show intern
[ Context 'intern' created by 'pbx_config' ]
  '210' =>          1. Dial(SIP/BT201)
[pbx_config]
  '211' =>          1. Dial(SIP/GXP1200)
[pbx_config]


I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.

I would love to have your feedback on this. Where could this problem be
situated ?

I notice (on the Asterisk CLI) that my SIP-phones do not register. They
have a fixed IP and there account information is set via the web
interface.

Greetingz,
Jonas.
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jonas kellens wrote:
> Hi there,
>
> this is the first time that I'm building an Asterisk-server.
>
> I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
> Zaptel is for later, when configuring the POTS-line. Now first 
> internal communication with SIP.
>
> Thought it would go easier...
>
> I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
>
> These are my settings :
>
> sip.conf :
> /[root at asterisk asterisk]# cat sip.conf/
> /[general]/
> /bindport=5060/
> /bindaddr = 0.0.0.0/
>
> /[BT201]/
> /type=friend/
> /context=intern/
> /host=192.168.4.210/
> /secret=testpaswoord/
>
> /[GXP1200]/
> /type=friend/
> /context=intern/
> /host=192.168.4.211/
> /secret=testpaswoord/
> extensions.conf :
> /[root at asterisk asterisk]# cat extensions.conf/
> /[intern]/
> /exten => 210,1,Dial(SIP/BT201)/
> /exten => 211,1,Dial(SIP/GXP1200)/
> Asterisk CLI shows me :
> /asterisk*CLI> sip reload/
> /Reloading SIP/
> /  == Parsing '/etc/asterisk/sip.conf': Found/
> /  == Parsing '/etc/asterisk/users.conf': Found/
> /  == Parsing '/etc/asterisk/sip_notify.conf': Found/
> /asterisk*CLI> sip show peers/
> /Name/username              Host            Dyn Nat ACL Port    
>  Status               /
> /GXP1200                    192.168.4.211               5060    
>  Unmonitored           /
> /BT201                      192.168.4.210               5060    
>  Unmonitored           /
> /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 
> offline]/
>
> /asterisk*CLI> dialplan show intern/
> /[ Context 'intern' created by 'pbx_config' ]/
> /  '210' =>          1. Dial(SIP/BT201)
> [pbx_config]/
> /  '211' =>          1. Dial(SIP/GXP1200)
> [pbx_config]/
>
> I pick up the phone of the BT201 and dial 211... nothing happens.
> I pick up the phone of the GXP1200 and dial 210... nothing happens.
>
> I would love to have your feedback on this. Where could this problem 
> be situated ?
>
> I notice (on the Asterisk CLI) that my SIP-phones do not register. 
> They have a fixed IP and there account information is set via the web 
> interface.
>
> Greetingz,
> Jonas.
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
I had the same issue.  I set the hosts to dynamic and and explicitly set 
their IP's via a dhcp server using their MAC addresses.  The phones 
registered and all is well.

Regards,
Mike

> bindaddr = 0.0.0.0
>
I would set this to the ethernet interface IP address, I believe this may be
your issue.

Registration is only for receiving calls, if you are not seeing information on
the dial, then the phone is not talking to the server.  I would make sure of the
settings in the web-interface as well.

Tony Plack

In your sip.conf or sip_nat.conf (for elastix) set the variables:

externhostexternipdomainexternrefreshlocalnet
Regards

Francisco



Anthony Plack wrote:
>> bindaddr = 0.0.0.0
>>
>>     
>
> I would set this to the ethernet interface IP address, I believe this may
be your issue.
>
> Registration is only for receiving calls, if you are not seeing information
on the dial, then the phone is not talking to the server.  I would make sure of
the settings in the web-interface as well.
>
> Tony Plack
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

On Mon, 13 Apr 2009, Anthony Plack wrote:
>> bindaddr = 0.0.0.0
>
> I would set this to the ethernet interface IP address, I believe this 
> may be your issue.
Binding to 0.0.0.0 means "listen to all IP addresses on the box." It
is
not the issue.

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000

What do you see when you run asterisk ?r and dial 210 or 211 from one of the
phones

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

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"Common sense is the collection of prejudices acquired by age
eighteen." -- Albert Einstein

"Once you can accept the universe as matter expanding into nothing that is
something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls
usingAsterisk

 

Hi there,

this is the first time that I'm building an Asterisk-server.

I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.

Thought it would go easier...

I have 2 Grandstream IP-phones : BT-201 and GXP-1200.

These are my settings :

sip.conf : 
[root at asterisk asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr = 0.0.0.0

[BT201]
type=friend
context=intern
host=192.168.4.210
secret=testpaswoord

[GXP1200]
type=friend
context=intern
host=192.168.4.211
secret=testpaswoord 
extensions.conf : 
[root at asterisk asterisk]# cat extensions.conf
[intern]
exten => 210,1,Dial(SIP/BT201)
exten => 211,1,Dial(SIP/GXP1200) 
Asterisk CLI shows me : 
asterisk*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
GXP1200                    192.168.4.211               5060     Unmonitored
BT201                      192.168.4.210               5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

asterisk*CLI> dialplan show intern
[ Context 'intern' created by 'pbx_config' ]
  '210' =>          1. Dial(SIP/BT201)                           
[pbx_config]
  '211' =>          1. Dial(SIP/GXP1200)                         
[pbx_config]

I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.

I would love to have your feedback on this. Where could this problem be situated
?

I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a
fixed IP and there account information is set via the web interface.

Greetingz,
Jonas. 

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jonas kellens wrote:
> I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI :
> 
> /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/
> /Verbosity is at least 5/
> /asterisk*CLI> /
> 
> Nothing is displayed... it stays that way...
> 
> Jonas.
Is there a "Send" button on that phone?  It sounds to me as though the
phone is still waiting for more digits.

Barry
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jonas kellens wrote:
> Hi there,
>
> I notice (on the Asterisk CLI) that my SIP-phones do not register. 
> They have a fixed IP and there account information is 
If your phones don't register, then your not going to be able to make a 
call. 

The Grandstream phones have a web interface (At least if memory serves 
correctly) and you'll need to tell the phones:

server ip address
username/extension to use
password of that extension.

Once they've registered, things will get a lot easier.

Doug



-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."

Hi

On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:
> I pick up the phone of the BT201 and dial 211... nothing happens.
> I pick up the phone of the GXP1200 and dial 210... nothing happens.
> 
> I would love to have your feedback on this. Where could this problem be
> situated ?
Your basic mistake at troubleshooting this is trying to test two things
at the same time. Let's test them separately.

1. A call from Asterisk to the phones:


In the Asterisk CLI:

  originate SIP/BT201 application playback demo-instruct

And the other one:

  originate SIP/GXP1200 application playback demo-instruct

Alternatively, use the echo-test aplication:

  originate SIP/BT201 application echo


2. Next, test calling from the phones to Asterisk. Add those two extensions
to [intern]

exten => 250,1,Answer
exten => 250,n,Playback(demo-instruct)
exten => 250,n,Hangup

exten => 251,1,Answer
exten => 251,1,Echo
exten => 251,1,Hangup

Make sure you reload for that to take effect, and then try dialing 250
or 251.

Another useful tools: 'sip debug'. It tends to generate a very noisy 
output that is normally not readable for mere mortals. However it does 
indicate that "something is happening". If you call from a remote SIP 
phone and there's nothing on the SIP debug, the problem is probably with 
the settings of the phone, as it is not getting to you.

Last and not least: a sanity check as you "see nothing": what is the
output of: 'logger show channels' ?

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir

Put register=yes in the BT201 and GXP1200 contexts of sip.conf

 

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls
usingAsterisk

 

Hi there,

this is the first time that I'm building an Asterisk-server.

I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.

Thought it would go easier...

I have 2 Grandstream IP-phones : BT-201 and GXP-1200.

These are my settings :

sip.conf : 
[root at asterisk asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr = 0.0.0.0

[BT201]
type=friend
context=intern
host=192.168.4.210
secret=testpaswoord

[GXP1200]
type=friend
context=intern
host=192.168.4.211
secret=testpaswoord 
extensions.conf : 
[root at asterisk asterisk]# cat extensions.conf
[intern]
exten => 210,1,Dial(SIP/BT201)
exten => 211,1,Dial(SIP/GXP1200) 
Asterisk CLI shows me : 
asterisk*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status

GXP1200                    192.168.4.211               5060     Unmonitored

BT201                      192.168.4.210               5060     Unmonitored

2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]

asterisk*CLI> dialplan show intern
[ Context 'intern' created by 'pbx_config' ]
  '210' =>          1. Dial(SIP/BT201)
[pbx_config]
  '211' =>          1. Dial(SIP/GXP1200)
[pbx_config] 

I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.

I would love to have your feedback on this. Where could this problem be
situated ?

I notice (on the Asterisk CLI) that my SIP-phones do not register. They have
a fixed IP and there account information is set via the web interface.

Greetingz,
Jonas. 

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just for a test, run "service iptables stop" as root on the asterisk 
server and then reboot your phones.  after that, try again and see if 
the phones are making communications with asterisk.

you can turn the firewall back on with "service iptables start"

jonas kellens wrote:
> Hi there,
> 
> this is the first time that I'm building an Asterisk-server.
> 
> I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
> Zaptel is for later, when configuring the POTS-line. Now first internal 
> communication with SIP.
> 
> Thought it would go easier...
> 
> I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
> 
> These are my settings :
> 
> sip.conf :
> /[root at asterisk asterisk]# cat sip.conf/
> /[general]/
> /bindport=5060/
> /bindaddr = 0.0.0.0/
> 
> /[BT201]/
> /type=friend/
> /context=intern/
> /host=192.168.4.210/
> /secret=testpaswoord/
> 
> /[GXP1200]/
> /type=friend/
> /context=intern/
> /host=192.168.4.211/
> /secret=testpaswoord/
> extensions.conf :
> /[root at asterisk asterisk]# cat extensions.conf/
> /[intern]/
> /exten => 210,1,Dial(SIP/BT201)/
> /exten => 211,1,Dial(SIP/GXP1200)/
> Asterisk CLI shows me :
> /asterisk*CLI> sip reload/
> /Reloading SIP/
> /  == Parsing '/etc/asterisk/sip.conf': Found/
> /  == Parsing '/etc/asterisk/users.conf': Found/
> /  == Parsing '/etc/asterisk/sip_notify.conf': Found/
> /asterisk*CLI> sip show peers/
> /Name/username              Host            Dyn Nat ACL Port     Status  
>              /
> /GXP1200                    192.168.4.211               5060    
>  Unmonitored           /
> /BT201                      192.168.4.210               5060    
>  Unmonitored           /
> /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 
> offline]/
> 
> /asterisk*CLI> dialplan show intern/
> /[ Context 'intern' created by 'pbx_config' ]/
> /  '210' =>          1. Dial(SIP/BT201)
> [pbx_config]/
> /  '211' =>          1. Dial(SIP/GXP1200)
> [pbx_config]/
> 
> I pick up the phone of the BT201 and dial 211... nothing happens.
> I pick up the phone of the GXP1200 and dial 210... nothing happens.
> 
> I would love to have your feedback on this. Where could this problem be 
> situated ?
> 
> I notice (on the Asterisk CLI) that my SIP-phones do not register. They 
> have a fixed IP and there account information is set via the web interface.
> 
> Greetingz,
> Jonas.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users